Chan_sip.c: Retransmission Timeout Reached On Transmission


Hello friends:

I am facing cutoffs randomly when negotiating calls.

The PBX dials the destination, the provider (softswitch) receives the request *[1]* and sudenly the PBX hangs up the call* [2]* while the provider is still dialing it, as a consequence the remote peer receives a ghost call. Along the atempt I could see six times a messages regarding NAT isuues *[3]*

I hope anyone can give me an idea to solve this issue. Softswitch is using an implementation of RFC 3264 and the PBX being used is Elastix 2.3 with Asterisk

Thanks in advance

Elder D. Arohuanca Lima - Peru

*[1]* [Aug 12…

Asterisk Users 2 months ago 3 Answers

Asterisk 13 FAX


Hello team! I’m planning to add fax functionality to my PBX. From research it seems that there is 2 options: spandsp and Digium. I lean towards Digium app, licensing is fine. However, they don’t have download for v13 Should I just download their version for v12 Asterisk? Any other suggestions on what to use, what works best? I have a pretty good plan on what I’m going to do but unsure which one to pick. Callcentric is my SIP provider and it supports T.38 Thanks Ivan --

Asterisk Users 4 months ago 2 Answers

Small Homebrew Pbx


Hello all,

I'm new here and I'm interested in building a small PBX with asterisk at home. I have one single PSTN line and ethernet cabling in place. I already have fairly decent PC that I can use (AMD FX 8350 16GB of RAM and RAID 10 SATA disks). I make and receive 10 calls a day on average. I want 4 IP phones connected to the ethernet network. When there is a incoming call, all phones must ring and the first that takes the call makes the others stop ringing, but lets them available for internal calls.

Given the requirements above,…

Asterisk Users 4 months ago 9 Answers

Asterisk Proxying A REFER


Hello, we are using Asterisk with Adhearsion as our application server, with another Asterisk box acting as the office PBX, where all office phones are registered.

A REFER to transfer calls within the office results in the Adhearsion application call being dropped, because the leg between the PBX and the app server is terminated by the PBX following the REFER. Is there a way to configure Asterisk 11 to proxy a refer across a bridge instead of following it, so the application server can follow it instead?

Best regards,

Luca Pradovera

Asterisk Users 6 months ago 2 Answers

Asterisk 13. Writing Call Quality Parameters To CDR. How?



Voice quality when calling - this is one of the most important in the PBX. You need to record the quality parameters for each call to improve.

Because the overall quality of a call can only be determined upon completion, I did it in the HangUp handler and wrote in custom fields of CDR. This worked well in asterisk 11.

In asterisk 13 I did not find a handler after the call, but before finalizing the CDR. I tried to call the AGI and there to update the CDR record by unique identifiers. But faced with the fact that there are no…

Asterisk Users 7 months ago 2 Answers